Hi,
I had some free time recently, so I had written some code for FFmpeg :)
I improved significantly DCA encoder. It works for any SUB(SUB)FRAMES and any
sample rate. Also files are played correctly with libdts. I will try to make
LFE and some extensions works soon.
--
Bartlomiej Wolowiec
Index: dcaenc.c
===================================================================
--- dcaenc.c (wersja 4178)
+++ dcaenc.c (kopia robocza)
@@ -23,19 +23,27 @@
#include "avcodec.h"
#include "bitstream.h"
#include "dcaenc.h"
+#include "dcadata.h"
-#define MAX_CHANNELS (2)
+#define MAX_CHANNELS (6)
#define DCA_SUBBANDS_32 (32)
#define DCA_MAX_FRAME_SIZE (16383)
-#define FIXED_FRAME_SIZE (7167)
+#define DCA_HEADER_SIZE 13
+#define SUBFRAMES 2
+#define SUBSUBFRAMES 2
+#define PCM_SAMPLES (SUBFRAMES*SUBSUBFRAMES*8)
+
typedef struct {
PutBitContext pb;
int32_t history[MAX_CHANNELS][512]; /* This is a circular buffer */
int start[MAX_CHANNELS];
+ int frame_size;
+ int prim_channels;
+ int sample_rate_code;
int32_t pcm[DCA_SUBBANDS_32];
- int32_t subband[64][MAX_CHANNELS][DCA_SUBBANDS_32]; /* [sample][channel][subband] */
+ int32_t subband[PCM_SAMPLES][MAX_CHANNELS][DCA_SUBBANDS_32]; /* [sample][channel][subband] */
} DCAContext;
static int32_t cos_table[128];
@@ -134,21 +142,19 @@
/* CRC is not present */
put_bits(&c->pb, 1, 0);
- /* Number of PCM sample blocks: 64
- (larger values are unusable with 1:1 compression due to high bitrate
- and frame size limitation) */
- put_bits(&c->pb, 7, 63);
+ /* Number of PCM sample blocks */
+ put_bits(&c->pb, 7, PCM_SAMPLES-1);
/* Primary frame byte size: 7168 */
- put_bits(&c->pb, 14, FIXED_FRAME_SIZE);
+ put_bits(&c->pb, 14, c->frame_size-1);
/* Audio channel arrangement: L + R (stereo) */
- put_bits(&c->pb, 6, 2);
+ put_bits(&c->pb, 6, c->prim_channels==2?2:9); //FIXME
- /* Core audio sampling frequency: 44100 Hz */
- put_bits(&c->pb, 4, 8);
+ /* Core audio sampling frequency */
+ put_bits(&c->pb, 4, c->sample_rate_code);
- /* Transmission bit rate: 1411.2 kbps */
+ /* Transmission bit rate: 1411.2 kbps */ //FIXME
put_bits(&c->pb, 5, 0x16);
/* Embedded down mix: disabled */
@@ -206,49 +212,56 @@
static void put_primary_audio_header(DCAContext *c)
{
- /* Number of subframes: 2 */
- put_bits(&c->pb, 4, 1);
+ /* From dca.c */
+ static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 };
+ static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };
- /* Number of primary audio channels: 2 */
- put_bits(&c->pb, 3, 1);
+ int ch, i;
+ /* Number of subframes */
+ put_bits(&c->pb, 4, SUBFRAMES-1);
+ /* Number of primary audio channels */
+ put_bits(&c->pb, 3, c->prim_channels-1);
+
/* Subband activity count: 27 + 27 */
- put_bits(&c->pb, 5, 25);
- put_bits(&c->pb, 5, 25);
+ for(ch=0; ch<c->prim_channels; ch++){
+ put_bits(&c->pb, 5, 25);
+ }
/* High frequency VQ start subband: 27, 27 */
- put_bits(&c->pb, 5, 26);
- put_bits(&c->pb, 5, 26);
+ for(ch=0; ch<c->prim_channels; ch++){
+ put_bits(&c->pb, 5, 26);
+ }
/* Joint intensity coding index: 0, 0 */
- put_bits(&c->pb, 3, 0);
- put_bits(&c->pb, 3, 0);
+ for(ch=0; ch<c->prim_channels; ch++){
+ put_bits(&c->pb, 3, 0);
+ }
/* Transient mode codebook: A4, A4 (arbitrary) */
- put_bits(&c->pb, 2, 0);
- put_bits(&c->pb, 2, 0);
+ for(ch=0; ch<c->prim_channels; ch++){
+ put_bits(&c->pb, 2, 0);
+ }
/* Scale factor code book: 7 bit linear, 7-bit sqrt table (for each channel) */
- put_bits(&c->pb, 3, 6);
- put_bits(&c->pb, 3, 6);
+ for(ch=0; ch<c->prim_channels; ch++){
+ put_bits(&c->pb, 3, 6);
+ }
/* Bit allocation quantizer select: linear 5-bit */
- put_bits(&c->pb, 3, 6);
- put_bits(&c->pb, 3, 6);
+ for(ch=0; ch<c->prim_channels; ch++){
+ put_bits(&c->pb, 3, 6);
+ }
/* Quantization index codebook select: dummy data
to avoid transmission of scale factor adjustment */
- put_bits(&c->pb, 1, 1); put_bits(&c->pb, 1, 1);
- put_bits(&c->pb, 2, 3); put_bits(&c->pb, 2, 3);
- put_bits(&c->pb, 2, 3); put_bits(&c->pb, 2, 3);
- put_bits(&c->pb, 2, 3); put_bits(&c->pb, 2, 3);
- put_bits(&c->pb, 2, 3); put_bits(&c->pb, 2, 3);
- put_bits(&c->pb, 3, 7); put_bits(&c->pb, 3, 7);
- put_bits(&c->pb, 3, 7); put_bits(&c->pb, 3, 7);
- put_bits(&c->pb, 3, 7); put_bits(&c->pb, 3, 7);
- put_bits(&c->pb, 3, 7); put_bits(&c->pb, 3, 7);
- put_bits(&c->pb, 3, 7); put_bits(&c->pb, 3, 7);
+ for(i=1; i<11; i++){
+ for(ch=0; ch<c->prim_channels; ch++){
+ put_bits(&c->pb, bitlen[i], thr[i]);
+ }
+ }
+
/* Scale factor adjustment index: not transmitted */
}
@@ -260,35 +273,36 @@
}
-static void put_subframe(DCAContext *c, int32_t subband_data[32][2][32])
+static void put_subframe(DCAContext *c, int32_t subband_data[8*SUBSUBFRAMES][MAX_CHANNELS][32])
{
int i, sub, ss, ch;
+ /* Subsubframes count */
+ put_bits(&c->pb, 2, SUBSUBFRAMES -1);
- /* Subsubframes count: 4 */
- put_bits(&c->pb, 2, 3);
-
/* Partial subsubframe sample count: dummy */
put_bits(&c->pb, 3, 0);
/* Prediction mode: no ADPCM, in each channel and subband */
- for (ch = 0; ch < 2; ch++)
+ for (ch = 0; ch < c->prim_channels; ch++)
for (sub = 0; sub < 27; sub++)
put_bits(&c->pb, 1, 0);
/* Prediction VQ addres: not transmitted */
/* Bit allocation index: 19 = "16 bits", for each channel and subband */
- for (ch = 0; ch < 2; ch++)
+ for (ch = 0; ch < c->prim_channels; ch++)
for (sub = 0; sub < 27; sub++)
put_bits(&c->pb, 5, 19);
- /* Transition mode: none for each channel and subband */
- for (ch = 0; ch < 2; ch++)
- for (sub = 0; sub < 27; sub++)
- put_bits(&c->pb, 1, 0); /* according to Huffman codebook A4 */
+ if(SUBSUBFRAMES>1){
+ /* Transition mode: none for each channel and subband */
+ for (ch = 0; ch < c->prim_channels; ch++)
+ for (sub = 0; sub < 27; sub++)
+ put_bits(&c->pb, 1, 0); /* according to Huffman codebook A4 */
+ }
/* Scale factors: the same for each channel and subband,
encoded according to Table D.1.2 */
- for (ch = 0; ch < 2; ch++)
+ for (ch = 0; ch < c->prim_channels; ch++)
for (sub = 0; sub < 27; sub++)
put_bits(&c->pb, 7, 110);
@@ -301,8 +315,8 @@
/* LFE data: none */
/* Audio data: 4 subsubframes */
- for (ss = 0; ss < 4 ; ss++)
- for (ch = 0; ch < 2; ch++)
+ for (ss = 0; ss < SUBSUBFRAMES ; ss++)
+ for (ch = 0; ch < c->prim_channels; ch++)
for (sub = 0; sub < 27; sub++)
for (i = 0; i < 8; i++)
put_bits(&c->pb, 16, quantize(subband_data[ss * 8 + i][ch][sub]));
@@ -310,17 +324,21 @@
put_bits(&c->pb, 16, 0xffff);
}
-void put_frame(DCAContext *c, int32_t subband_data[64][2][32], uint8_t *frame)
+void put_frame(DCAContext *c, int32_t subband_data[PCM_SAMPLES][MAX_CHANNELS][32], uint8_t *frame)
{
- int channel;
- init_put_bits(&c->pb, frame, DCA_MAX_FRAME_SIZE);
+ int i;
+ init_put_bits(&c->pb, frame + DCA_HEADER_SIZE, DCA_MAX_FRAME_SIZE-DCA_HEADER_SIZE);
- put_frame_header(c);
put_primary_audio_header(c);
- for (channel=0 ; channel<2; channel++)
- put_subframe(c, &subband_data[32 * channel]);
+ for(i=0; i<SUBFRAMES; i++)
+ put_subframe(c, &subband_data[SUBSUBFRAMES * 8 * i]);
flush_put_bits(&c->pb);
+ c->frame_size = (put_bits_count(&c->pb)>>3) + DCA_HEADER_SIZE;
+
+ init_put_bits(&c->pb, frame, DCA_HEADER_SIZE);
+ put_frame_header(c);
+ flush_put_bits(&c->pb);
}
static int DCA_encode_frame(AVCodecContext *avctx,
@@ -333,11 +351,11 @@
// if (buf_size < MAX_CHANNELS*2048*sizeof(int16_t))
// return -1;
- for (i = 0; i < 64; i ++) /* i is the decimated sample number */
- for (channel=0; channel<2 ; channel++) {
+ for (i = 0; i < PCM_SAMPLES; i ++) /* i is the decimated sample number */
+ for (channel=0; channel<c->prim_channels; channel++) {
/* Get 32 PCM samples */
for (k = 0; k < 32; k++) { /* k is the sample number in a 32-sample block */
- c->pcm[k] = samples[2 * (32*i+k) + channel] << 16;
+ c->pcm[k] = samples[avctx->channels * (32*i+k) + channel] << 16;
}
/* Put subband samples into the proper place */
qmf_decompose(c, c->pcm, &c->subband[i][channel][0], channel);
@@ -345,19 +363,32 @@
put_frame(c, c->subband, frame);
- return put_bits_count(&c->pb)>>3;
+ return c->frame_size;
}
static int DCA_encode_init(AVCodecContext *avctx) {
- //DCAContext *c = avctx->priv_data;
+ DCAContext *c = avctx->priv_data;
+ int i;
- if(avctx->channels != 2 || avctx->sample_rate != 44100) {
- av_log(avctx, AV_LOG_ERROR, "Only 44.1 kHz stereo is supported at the moment!\n");
+ c->prim_channels = FFMIN(avctx->channels, 5); //XXX only 5 channels
+
+ for(i=0; i<16; i++){
+ if(dca_sample_rates[i] == avctx->sample_rate)
+ break;
+ }
+ if(i==16){
+ av_log(avctx, AV_LOG_ERROR, "Sample rate %iHz not supported\n", avctx->sample_rate);
return -1;
}
+ c->sample_rate_code = i;
- avctx->frame_size = 2048;
+ if(avctx->channels != 2 && avctx->channels != 6) {
+ av_log(avctx, AV_LOG_ERROR, "Only stereo and 5.1 supported at the moment!\n");
+ return -1;
+ }
+ avctx->frame_size = 32 * PCM_SAMPLES;
+
qmf_init();
return 0;
}
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